Video Calls drops and oneway after 30 seconds

[latif81]

Hello, I have issue with video calls via webrtc, video call start working fine, after 5 or 10 sec video starts to stutter and lag, then after 25 or 30 seconds video call becomes one-way (audio working fine 2 ways and no error or warning in logs),
Here is my endpoint config:

[777]
type=endpoint
transport=transport-wss
context=stdexten
callerid=777 <777>
auth=777
aors=777
from_domain=sip.mydomain.com
dtls_cert_file=/etc/asterisk/certificate.crt
dtls_private_key=/etc/asterisk/private.key
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=ulaw,alaw,gsm,g726,g722,h264,mpeg4
dtmf_mode=rfc4733
direct_media=yes
aggregate_mwi=yes
use_avpf=yes
rtcp_mux=yes
max_audio_streams=1
max_video_streams=1
bundle=yes
ice_support=yes
media_use_received_transport=no
trust_id_inbound=yes
user_eq_phone=no
send_connected_line=yes
media_encryption=dtls
timers=yes
timers_min_se=90
media_encryption_optimistic=no
refer_blind_progress=yes
refer_blind_progress=yes
rtp_timeout=600
rtp_timeout_hold=600
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord
dtls_verify=fingerprint
dtls_setup=actpass
dtls_rekey=0

any advice?
Thanks

[InnovateAsterisk]

try rtp set debug on to start debugging your RTP packet flow… you should see clearly the missing packets, then trace them back.

Note: this command line dumps a lot of lines onto the screen, if you are on a production box… this will be an unreadable amount - just do this on a test box with 1 call. Then use rtp set debug off when you are done.

If audio is fine and video breaks up it could be to do with bandwidth restrictions - obviously video take up more than audio only.

Then again it could be replay check failed (index too old).

[latif81]

Thanks for your reply, I noticed that this issue only occurs with Google Chrome, even though it has been updated to the latest version (normal and private)